1.1.0b - test prerelease.[..]* Make less global gain against overloading.

I don't think that was necessary in foobar2000, as in the DSP pipeline 64 bit floats are used, it won't overload easily. And now the volume is a little bit lower with the b2bs plugin than without.Nevertheless, for me this is the best "headphone" plugin for foobar so far.

1.1.0b - test prerelease.[..]* Make less global gain against overloading.

I don't think that was necessary in foobar2000, as in the DSP pipeline 64 bit floats are used, it won't overload easily. And now the volume is a little bit lower with the b2bs plugin than without.Nevertheless, for me this is the best "headphone" plugin for foobar so far.

Thanks.Floating point sound stream must be normalized to 1.0 value peak. That is why "diskwriter" plugin can be overloaded by bs2b plugin when it writes to integer wav file. bs2b do not make normalization of stream and diskwriter does not too.I was add a little attenuation because a mathematical calculated global gain make overload in some frequency range with highs boost option.I saw this by equal left and right sine signals (mono). I was prefer to make less global gain for all cases than make it due to bs2b options because this algebra is so difficult to me :-( and I think that this task have no solution.

thank you so much for writing bs2b. I used to listen with headphones a lot, and always wanted to have a decent crossfeed plugin. The one built into foobar is not very good. I downloaded your plugin, and now I use it all the time. In particular, I am delighted how symphonic and opera music sounds. The soundstage is now very clear, the music is less fatiguing, the bass is sitting very well in the overall mix. I started to recommend the plugin on some forums and I was quite surprised with sometimes not very enthusiastic responses. They were however caused by the gain reduction which you mentioned recently. People check the effects of the plugin by removing it from the DSP list and putting back while playing. What they notice immediately is that the music with bs2b is more quiet and they complain that dynamics is reduced. This is silly, and people who really can listen will appreciate the plugin.But if you want the plugin to win popularity, I encourage you to tweak it in such way, that gain is not reduced.

New 2.0.0b version of bs2b plug-in was released.Sources and win32 binaries are available at http://bs2b.sourceforge.net/Release notes: * The new high frequency boost filter have implemented. The old version of high-boost filter have been realized by two-step recursive filter for computation power conserving by subtraction of low-pass filter signal. The new method is a one-step recursive filter. It has done to provide an adjusted cut frequency value for more smooth frequency responce of resulting signal. * The new clipping feature by checking of [-1, +1] range of double float operations have implemented against possible overloads of signal level. * Global gain have calculated from levels of low frequency range like in first release. This is don't makes overload now due to the new frequency responce and to the new clipping feature. * New functions have added to library for various integer audio data processing. * New tuning method have implemented by three preset levels. * Updatted Winamp 2 plugin have included to package.

I'm trying to understand this new version. The old version allowed you to choose either Moy's or Linkwitz' crossfeed level. It also allowed you to choose a hgih boost or not.The only control with this version is a three position slider for crossfeed level. What does each position equate to. Also, is the high boost built in?

Sorry for not understanding this but to my ears this is by far the best crossfeed plugin available so I'd like to continue using it.

Whoa, I don't fully understand how this works, or really what it does, but after using it, listening without it is almost unbearable. "Getting the sound out of your head" is a very accurate description of its effect.

I'm trying to understand this new version. The old version allowed you to choose either Moy's or Linkwitz' crossfeed level. It also allowed you to choose a hgih boost or not.The only control with this version is a three position slider for crossfeed level. What does each position equate to. Also, is the high boost built in?

The old version of bs2b, I was found, is not realy like Moy's or Linkwitz's versions. Because my old high-boost filter have more low cut frequensy. This is mistake of my mind that leads to more thick lowmids. The new version of bs2b is like Moy's version but not exactly.I have done draft document. Please, see it at http://bs2b.sourceforge.net/about-draft.htmlI will report to forum about release of comlete new manual.

And there is SO MUCH in the music you dont normally hear. I just listened to Hey Jude from my Past Masters Vol 2 cd (for those of you who dont know, this is a compilation of singles) and --->

-->when listening at 3:14 you can make out the words "make it jude" and "jude jude jude wawawawawawaw" and all of the other stuff which were from alternate takes which you can hear in the anthology. You can see how the overlayed the tracks with it and its really cool! So much subtlty is exposed with this plugin - and it sounds great!Ill try and find more examples as I come upon them.

i dont understand. what am i supposed to do with the (.9) plug-in? when i add it to the playback DSP, all it does is muffle the music. it sounds like it turns down the mid-level freqs in the EQ, and the config options don't make a perceivable difference. am i supposed to have a certain kind of headphones or something?

Uh, I too need to know how to apply resampling (I resample to 48000Khz instead of letting my hardware do it) if this also resamples to 41Khz ? Should this go before resampling or after ?

In case you're unaware, the reason why people resample in foobar to 48000 Khz is because their hardware automatically does it, and foobar's resampler would do a better job. But I don't want to be having dsp resampling hell (41 Khz > 48 Khz(DSP) > 41 Khz (Baeur DSP) > 48Khz (Soundcard) ) would be bad for example.

Does that mean it is not appropriate to pass 48000 Hz audio to this DSP? Currently I am placing the resampler in front of bs2b...

It is mean that a user of bs2b library can bypass initialisation calls until he need to change crossfeed level or change sample rate in order to rate of audio samples.It is no concern of plugin or converter users.